Mercurial > libavcodec.hg
annotate resample.c @ 1909:559e92baf533 libavcodec
fix decoding of qp<6
author | michael |
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date | Fri, 26 Mar 2004 20:28:29 +0000 |
parents | 932d306bf1dc |
children | 3dc9bbe1b152 |
rev | line source |
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0 | 1 /* |
2 * Sample rate convertion for both audio and video | |
429 | 3 * Copyright (c) 2000 Fabrice Bellard. |
0 | 4 * |
429 | 5 * This library is free software; you can redistribute it and/or |
6 * modify it under the terms of the GNU Lesser General Public | |
7 * License as published by the Free Software Foundation; either | |
8 * version 2 of the License, or (at your option) any later version. | |
0 | 9 * |
429 | 10 * This library is distributed in the hope that it will be useful, |
0 | 11 * but WITHOUT ANY WARRANTY; without even the implied warranty of |
429 | 12 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU |
13 * Lesser General Public License for more details. | |
0 | 14 * |
429 | 15 * You should have received a copy of the GNU Lesser General Public |
16 * License along with this library; if not, write to the Free Software | |
17 * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA | |
0 | 18 */ |
1106 | 19 |
20 /** | |
21 * @file resample.c | |
22 * Sample rate convertion for both audio and video. | |
23 */ | |
24 | |
64 | 25 #include "avcodec.h" |
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26 |
0 | 27 typedef struct { |
28 /* fractional resampling */ | |
1064 | 29 uint32_t incr; /* fractional increment */ |
30 uint32_t frac; | |
0 | 31 int last_sample; |
32 /* integer down sample */ | |
33 int iratio; /* integer divison ratio */ | |
34 int icount, isum; | |
35 int inv; | |
36 } ReSampleChannelContext; | |
37 | |
38 struct ReSampleContext { | |
39 ReSampleChannelContext channel_ctx[2]; | |
40 float ratio; | |
41 /* channel convert */ | |
42 int input_channels, output_channels, filter_channels; | |
43 }; | |
44 | |
45 | |
46 #define FRAC_BITS 16 | |
47 #define FRAC (1 << FRAC_BITS) | |
48 | |
49 static void init_mono_resample(ReSampleChannelContext *s, float ratio) | |
50 { | |
51 ratio = 1.0 / ratio; | |
1057 | 52 s->iratio = (int)floorf(ratio); |
0 | 53 if (s->iratio == 0) |
54 s->iratio = 1; | |
55 s->incr = (int)((ratio / s->iratio) * FRAC); | |
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56 s->frac = FRAC; |
0 | 57 s->last_sample = 0; |
58 s->icount = s->iratio; | |
59 s->isum = 0; | |
60 s->inv = (FRAC / s->iratio); | |
61 } | |
62 | |
63 /* fractional audio resampling */ | |
64 static int fractional_resample(ReSampleChannelContext *s, short *output, short *input, int nb_samples) | |
65 { | |
66 unsigned int frac, incr; | |
67 int l0, l1; | |
68 short *q, *p, *pend; | |
69 | |
70 l0 = s->last_sample; | |
71 incr = s->incr; | |
72 frac = s->frac; | |
73 | |
74 p = input; | |
75 pend = input + nb_samples; | |
76 q = output; | |
77 | |
78 l1 = *p++; | |
79 for(;;) { | |
80 /* interpolate */ | |
81 *q++ = (l0 * (FRAC - frac) + l1 * frac) >> FRAC_BITS; | |
82 frac = frac + s->incr; | |
83 while (frac >= FRAC) { | |
739 | 84 frac -= FRAC; |
0 | 85 if (p >= pend) |
86 goto the_end; | |
87 l0 = l1; | |
88 l1 = *p++; | |
89 } | |
90 } | |
91 the_end: | |
92 s->last_sample = l1; | |
93 s->frac = frac; | |
94 return q - output; | |
95 } | |
96 | |
97 static int integer_downsample(ReSampleChannelContext *s, short *output, short *input, int nb_samples) | |
98 { | |
99 short *q, *p, *pend; | |
100 int c, sum; | |
101 | |
102 p = input; | |
103 pend = input + nb_samples; | |
104 q = output; | |
105 | |
106 c = s->icount; | |
107 sum = s->isum; | |
108 | |
109 for(;;) { | |
110 sum += *p++; | |
111 if (--c == 0) { | |
112 *q++ = (sum * s->inv) >> FRAC_BITS; | |
113 c = s->iratio; | |
114 sum = 0; | |
115 } | |
116 if (p >= pend) | |
117 break; | |
118 } | |
119 s->isum = sum; | |
120 s->icount = c; | |
121 return q - output; | |
122 } | |
123 | |
124 /* n1: number of samples */ | |
125 static void stereo_to_mono(short *output, short *input, int n1) | |
126 { | |
127 short *p, *q; | |
128 int n = n1; | |
129 | |
130 p = input; | |
131 q = output; | |
132 while (n >= 4) { | |
133 q[0] = (p[0] + p[1]) >> 1; | |
134 q[1] = (p[2] + p[3]) >> 1; | |
135 q[2] = (p[4] + p[5]) >> 1; | |
136 q[3] = (p[6] + p[7]) >> 1; | |
137 q += 4; | |
138 p += 8; | |
139 n -= 4; | |
140 } | |
141 while (n > 0) { | |
142 q[0] = (p[0] + p[1]) >> 1; | |
143 q++; | |
144 p += 2; | |
145 n--; | |
146 } | |
147 } | |
148 | |
149 /* n1: number of samples */ | |
150 static void mono_to_stereo(short *output, short *input, int n1) | |
151 { | |
152 short *p, *q; | |
153 int n = n1; | |
154 int v; | |
155 | |
156 p = input; | |
157 q = output; | |
158 while (n >= 4) { | |
159 v = p[0]; q[0] = v; q[1] = v; | |
160 v = p[1]; q[2] = v; q[3] = v; | |
161 v = p[2]; q[4] = v; q[5] = v; | |
162 v = p[3]; q[6] = v; q[7] = v; | |
163 q += 8; | |
164 p += 4; | |
165 n -= 4; | |
166 } | |
167 while (n > 0) { | |
168 v = p[0]; q[0] = v; q[1] = v; | |
169 q += 2; | |
170 p += 1; | |
171 n--; | |
172 } | |
173 } | |
174 | |
175 /* XXX: should use more abstract 'N' channels system */ | |
176 static void stereo_split(short *output1, short *output2, short *input, int n) | |
177 { | |
178 int i; | |
179 | |
180 for(i=0;i<n;i++) { | |
181 *output1++ = *input++; | |
182 *output2++ = *input++; | |
183 } | |
184 } | |
185 | |
186 static void stereo_mux(short *output, short *input1, short *input2, int n) | |
187 { | |
188 int i; | |
189 | |
190 for(i=0;i<n;i++) { | |
191 *output++ = *input1++; | |
192 *output++ = *input2++; | |
193 } | |
194 } | |
195 | |
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196 static void ac3_5p1_mux(short *output, short *input1, short *input2, int n) |
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197 { |
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198 int i; |
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199 short l,r; |
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200 |
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201 for(i=0;i<n;i++) { |
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202 l=*input1++; |
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203 r=*input2++; |
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204 *output++ = l; /* left */ |
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205 *output++ = (l/2)+(r/2); /* center */ |
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206 *output++ = r; /* right */ |
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207 *output++ = 0; /* left surround */ |
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208 *output++ = 0; /* right surroud */ |
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209 *output++ = 0; /* low freq */ |
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210 } |
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211 } |
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212 |
0 | 213 static int mono_resample(ReSampleChannelContext *s, short *output, short *input, int nb_samples) |
214 { | |
64 | 215 short *buf1; |
0 | 216 short *buftmp; |
217 | |
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218 buf1= (short*)av_malloc( nb_samples * sizeof(short) ); |
64 | 219 |
0 | 220 /* first downsample by an integer factor with averaging filter */ |
221 if (s->iratio > 1) { | |
222 buftmp = buf1; | |
223 nb_samples = integer_downsample(s, buftmp, input, nb_samples); | |
224 } else { | |
225 buftmp = input; | |
226 } | |
227 | |
228 /* then do a fractional resampling with linear interpolation */ | |
229 if (s->incr != FRAC) { | |
230 nb_samples = fractional_resample(s, output, buftmp, nb_samples); | |
231 } else { | |
232 memcpy(output, buftmp, nb_samples * sizeof(short)); | |
233 } | |
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234 av_free(buf1); |
0 | 235 return nb_samples; |
236 } | |
237 | |
238 ReSampleContext *audio_resample_init(int output_channels, int input_channels, | |
239 int output_rate, int input_rate) | |
240 { | |
241 ReSampleContext *s; | |
242 int i; | |
243 | |
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244 if ( input_channels > 2) |
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245 { |
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246 av_log(NULL, AV_LOG_ERROR, "Resampling with input channels greater than 2 unsupported."); |
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247 return NULL; |
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248 } |
0 | 249 |
250 s = av_mallocz(sizeof(ReSampleContext)); | |
251 if (!s) | |
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252 { |
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253 av_log(NULL, AV_LOG_ERROR, "Can't allocate memory for resample context."); |
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254 return NULL; |
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255 } |
0 | 256 |
257 s->ratio = (float)output_rate / (float)input_rate; | |
258 | |
259 s->input_channels = input_channels; | |
260 s->output_channels = output_channels; | |
261 | |
262 s->filter_channels = s->input_channels; | |
263 if (s->output_channels < s->filter_channels) | |
264 s->filter_channels = s->output_channels; | |
265 | |
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266 /* |
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267 * ac3 output is the only case where filter_channels could be greater than 2. |
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268 * input channels can't be greater than 2, so resample the 2 channels and then |
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269 * expand to 6 channels after the resampling. |
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270 */ |
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271 if(s->filter_channels>2) |
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272 s->filter_channels = 2; |
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273 |
0 | 274 for(i=0;i<s->filter_channels;i++) { |
275 init_mono_resample(&s->channel_ctx[i], s->ratio); | |
276 } | |
277 return s; | |
278 } | |
279 | |
280 /* resample audio. 'nb_samples' is the number of input samples */ | |
281 /* XXX: optimize it ! */ | |
282 /* XXX: do it with polyphase filters, since the quality here is | |
283 HORRIBLE. Return the number of samples available in output */ | |
284 int audio_resample(ReSampleContext *s, short *output, short *input, int nb_samples) | |
285 { | |
286 int i, nb_samples1; | |
64 | 287 short *bufin[2]; |
288 short *bufout[2]; | |
0 | 289 short *buftmp2[2], *buftmp3[2]; |
64 | 290 int lenout; |
0 | 291 |
292 if (s->input_channels == s->output_channels && s->ratio == 1.0) { | |
293 /* nothing to do */ | |
294 memcpy(output, input, nb_samples * s->input_channels * sizeof(short)); | |
295 return nb_samples; | |
296 } | |
297 | |
64 | 298 /* XXX: move those malloc to resample init code */ |
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299 bufin[0]= (short*) av_malloc( nb_samples * sizeof(short) ); |
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300 bufin[1]= (short*) av_malloc( nb_samples * sizeof(short) ); |
64 | 301 |
302 /* make some zoom to avoid round pb */ | |
303 lenout= (int)(nb_samples * s->ratio) + 16; | |
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304 bufout[0]= (short*) av_malloc( lenout * sizeof(short) ); |
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305 bufout[1]= (short*) av_malloc( lenout * sizeof(short) ); |
64 | 306 |
0 | 307 if (s->input_channels == 2 && |
308 s->output_channels == 1) { | |
309 buftmp2[0] = bufin[0]; | |
310 buftmp3[0] = output; | |
311 stereo_to_mono(buftmp2[0], input, nb_samples); | |
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312 } else if (s->output_channels >= 2 && s->input_channels == 1) { |
0 | 313 buftmp2[0] = input; |
314 buftmp3[0] = bufout[0]; | |
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315 } else if (s->output_channels >= 2) { |
0 | 316 buftmp2[0] = bufin[0]; |
317 buftmp2[1] = bufin[1]; | |
318 buftmp3[0] = bufout[0]; | |
319 buftmp3[1] = bufout[1]; | |
320 stereo_split(buftmp2[0], buftmp2[1], input, nb_samples); | |
321 } else { | |
322 buftmp2[0] = input; | |
323 buftmp3[0] = output; | |
324 } | |
325 | |
326 /* resample each channel */ | |
327 nb_samples1 = 0; /* avoid warning */ | |
328 for(i=0;i<s->filter_channels;i++) { | |
329 nb_samples1 = mono_resample(&s->channel_ctx[i], buftmp3[i], buftmp2[i], nb_samples); | |
330 } | |
331 | |
332 if (s->output_channels == 2 && s->input_channels == 1) { | |
333 mono_to_stereo(output, buftmp3[0], nb_samples1); | |
334 } else if (s->output_channels == 2) { | |
335 stereo_mux(output, buftmp3[0], buftmp3[1], nb_samples1); | |
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336 } else if (s->output_channels == 6) { |
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337 ac3_5p1_mux(output, buftmp3[0], buftmp3[1], nb_samples1); |
0 | 338 } |
339 | |
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340 av_free(bufin[0]); |
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341 av_free(bufin[1]); |
64 | 342 |
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343 av_free(bufout[0]); |
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344 av_free(bufout[1]); |
0 | 345 return nb_samples1; |
346 } | |
347 | |
348 void audio_resample_close(ReSampleContext *s) | |
349 { | |
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350 av_free(s); |
0 | 351 } |